- remove extra sip users
- rename a few properties
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@ -21,12 +21,9 @@ public class Application extends MultiThreadedApplicationAdapter {
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private ClientConnectionManager clientConnManager;
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private String sipServerHost = "localhost";
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private int startSipPort = 5070;
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private int stopSipPort = 5099;
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private int sipPort;
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private int startRtpPort = 3000;
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private int stopRtpPort = 3029;
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private int rtpPort;
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private int sipPort = 5070;
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private int startAudioPort = 3000;
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private int stopAudioPort = 3029;
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private String password = "secret";
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private String username;
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private CallStreamFactory callStreamFactory;
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@ -38,7 +35,7 @@ public class Application extends MultiThreadedApplicationAdapter {
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callStreamFactory.setScope(scope);
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sipPeerManager.setCallStreamFactory(callStreamFactory);
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sipPeerManager.setClientConnectionManager(clientConnManager);
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sipPeerManager.createSipPeer("default", sipServerHost, startSipPort, startRtpPort, stopRtpPort);
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sipPeerManager.createSipPeer("default", sipServerHost, sipPort, startAudioPort, stopAudioPort);
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try {
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sipPeerManager.register("default", username, password);
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} catch (PeerNotFoundException e) {
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@ -125,20 +122,16 @@ public class Application extends MultiThreadedApplicationAdapter {
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this.password = pw;
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}
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public void setStartSIPPort(int startSIPPort) {
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this.startSipPort = startSIPPort;
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public void setSipPort(int sipPort) {
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this.sipPort = sipPort;
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}
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public void setStopSIPPort(int stopSIPPort) {
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this.stopSipPort = stopSIPPort;
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public void setStartAudioPort(int startRTPPort) {
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this.startAudioPort = startRTPPort;
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}
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public void setStartRTPPort(int startRTPPort) {
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this.startRtpPort = startRTPPort;
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}
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public void setStopRTPPort(int stopRTPPort) {
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this.stopRtpPort = stopRTPPort;
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public void setStopAudioPort(int stopRTPPort) {
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this.stopAudioPort = stopRTPPort;
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}
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public void setSipPeerManager(SipPeerManager spm) {
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@ -75,6 +75,9 @@ public class SipPeer implements SipRegisterAgentListener {
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registeredProfile.keepaliveTime=8000;
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registeredProfile.acceptTime=0;
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registeredProfile.hangupTime=20;
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log.debug( "SIPUser register : {}", fromURL );
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log.debug( "SIPUser register : {}", registeredProfile.contactUrl );
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}
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@ -1,18 +1,10 @@
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# The address of your asterisk server
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#sip.server.host=192.168.0.182
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sip.server.host=127.0.0.1
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sip.server.username=3000
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sip.server.port=5070
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sip.server.username=bbbuser
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sip.server.password=secret
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# The start/stop SIP ports that the application is going
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# to use to connect to the asterisk server.
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# These are unassigned ports as defined here
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# (http://www.iana.org/assignments/port-numbers)
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# We are setting up 100 default SIP users (see RTP ports below)
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# so we need one port per user.
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startSIPPort=34480
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stopSIPPort=34579
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# The start/stop RTP port the application is going to use
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# for the media stream.
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# NOTE: This will also be used as SIP users to REGISTER with
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@ -20,8 +12,8 @@ stopSIPPort=34579
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# in your bbb_sip.conf.
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# See http://code.google.com/p/bigbluebutton/source/browse/#svn/trunk/bbb-voice-conference/config/asterisk
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# create-sip-users.sh script to create the users.
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startRTPPort=34380
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stopRTPPort=34479
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startAudioPort=34380
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stopAudioPort=34479
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# An extension pattern, in case your asterisk extensions.conf
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# uses a naming convetion for your meeting rooms
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@ -30,12 +30,11 @@
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<bean id="web.handler" class="org.bigbluebutton.voiceconf.red5.Application">
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<property name="sipServerHost" value="${sip.server.host}" />
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<property name="sipPort" value="${sip.server.port}" />
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<property name="username" value="${sip.server.username}" />
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<property name="password" value="${sip.server.password}" />
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<property name="startSIPPort" value="${startSIPPort}" />
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<property name="stopSIPPort" value="${stopSIPPort}" />
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<property name="startRTPPort" value="${startRTPPort}" />
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<property name="stopRTPPort" value="${stopRTPPort}" />
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<property name="startAudioPort" value="${startAudioPort}" />
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<property name="stopAudioPort" value="${stopAudioPort}" />
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<property name="sipPeerManager" ref="sipPeerManager"/>
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<property name="clientConnectionManager" ref="clientConnectionManager"/>
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</bean>
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